Extract most of PacedSender into PacedSendingController.
The Pacer now just handles interaction with Module/ProcessThread and forwarding packets to PacketRouter. All other logic is moved to PacedSendingController, including tests. PacedSender unittest are now just some basic sanity tests. Bug: webrtc:10809 Change-Id: I69223cd9d8300997375b03706d2e99c88e46241c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149041 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28886}
This commit is contained in:
Erik Språng
221
modules/pacing/pacing_controller.h
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modules/pacing/pacing_controller.h
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/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_PACING_PACING_CONTROLLER_H_
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#define MODULES_PACING_PACING_CONTROLLER_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <atomic>
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#include <memory>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/function_view.h"
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "api/transport/field_trial_based_config.h"
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#include "api/transport/network_types.h"
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#include "api/transport/webrtc_key_value_config.h"
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#include "modules/pacing/bitrate_prober.h"
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#include "modules/pacing/interval_budget.h"
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#include "modules/pacing/round_robin_packet_queue.h"
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#include "modules/pacing/rtp_packet_pacer.h"
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#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/experiments/field_trial_parser.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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// This class implements a leaky-buck packet pacing algorithm. It handles the
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// logic of determining which packets to send when, but the actual timing of
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// the processing is done externally (e.g. PacedSender). Furthermore, the
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// forwarding of packets when they are ready to be sent is also handled
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// externally, via the PacedSendingController::PacketSender interface.
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//
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class PacingController {
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public:
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class PacketSender {
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public:
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virtual ~PacketSender() = default;
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virtual void SendRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
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const PacedPacketInfo& cluster_info) = 0;
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virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
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DataSize size) = 0;
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// TODO(bugs.webrtc.org/10633): Remove these when old code path is gone.
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virtual RtpPacketSendResult TimeToSendPacket(
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uint32_t ssrc,
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uint16_t sequence_number,
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int64_t capture_timestamp,
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bool retransmission,
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const PacedPacketInfo& packet_info) = 0;
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virtual DataSize TimeToSendPadding(DataSize size,
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const PacedPacketInfo& pacing_info) = 0;
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};
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// Expected max pacer delay. If ExpectedQueueTime() is higher than
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// this value, the packet producers should wait (eg drop frames rather than
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// encoding them). Bitrate sent may temporarily exceed target set by
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// UpdateBitrate() so that this limit will be upheld.
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static const TimeDelta kMaxExpectedQueueLength;
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// Pacing-rate relative to our target send rate.
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// Multiplicative factor that is applied to the target bitrate to calculate
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// the number of bytes that can be transmitted per interval.
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// Increasing this factor will result in lower delays in cases of bitrate
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// overshoots from the encoder.
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static const float kDefaultPaceMultiplier;
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// If no media or paused, wake up at least every |kPausedProcessIntervalMs| in
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// order to send a keep-alive packet so we don't get stuck in a bad state due
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// to lack of feedback.
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static const TimeDelta kPausedProcessInterval;
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PacingController(Clock* clock,
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PacketSender* packet_sender,
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RtcEventLog* event_log,
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const WebRtcKeyValueConfig* field_trials);
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~PacingController();
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// Adds the packet information to the queue and calls TimeToSendPacket
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// when it's time to send.
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void InsertPacket(RtpPacketSender::Priority priority,
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uint32_t ssrc,
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uint16_t sequence_number,
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int64_t capture_time_ms,
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size_t bytes,
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bool retransmission);
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// Adds the packet to the queue and calls PacketRouter::SendPacket() when
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// it's time to send.
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void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet);
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void CreateProbeCluster(DataRate bitrate, int cluster_id);
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void Pause(); // Temporarily pause all sending.
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void Resume(); // Resume sending packets.
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bool IsPaused() const;
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void SetCongestionWindow(DataSize congestion_window_size);
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void UpdateOutstandingData(DataSize outstanding_data);
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// Sets the pacing rates. Must be called once before packets can be sent.
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void SetPacingRates(DataRate pacing_rate, DataRate padding_rate);
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// Currently audio traffic is not accounted by pacer and passed through.
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// With the introduction of audio BWE audio traffic will be accounted for
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// the pacer budget calculation. The audio traffic still will be injected
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// at high priority.
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void SetAccountForAudioPackets(bool account_for_audio);
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// Returns the time since the oldest queued packet was enqueued.
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TimeDelta OldestPacketWaitTime() const;
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size_t QueueSizePackets() const;
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DataSize QueueSizeData() const;
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// Returns the time when the first packet was sent;
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absl::optional<Timestamp> FirstSentPacketTime() const;
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// Returns the number of milliseconds it will take to send the current
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// packets in the queue, given the current size and bitrate, ignoring prio.
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TimeDelta ExpectedQueueTime() const;
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void SetQueueTimeLimit(TimeDelta limit);
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// Enable bitrate probing. Enabled by default, mostly here to simplify
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// testing. Must be called before any packets are being sent to have an
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// effect.
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void SetProbingEnabled(bool enabled);
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// Time until next probe should be sent. If this value is set, it should be
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// respected - i.e. don't call ProcessPackets() before this specified time as
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// that can have unintended side effects.
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absl::optional<TimeDelta> TimeUntilNextProbe();
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// Time since ProcessPackets() was last executed.
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TimeDelta TimeElapsedSinceLastProcess() const;
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TimeDelta TimeUntilAvailableBudget() const;
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// Check queue of pending packets and send them or padding packets, if budget
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// is available.
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void ProcessPackets();
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bool Congested() const;
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private:
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TimeDelta UpdateTimeAndGetElapsed(Timestamp now);
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bool ShouldSendKeepalive(Timestamp now) const;
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// Updates the number of bytes that can be sent for the next time interval.
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void UpdateBudgetWithElapsedTime(TimeDelta delta);
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void UpdateBudgetWithSentData(DataSize size);
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DataSize PaddingToAdd(absl::optional<DataSize> recommended_probe_size,
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DataSize data_sent);
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RoundRobinPacketQueue::QueuedPacket* GetPendingPacket(
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const PacedPacketInfo& pacing_info);
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void OnPacketSent(RoundRobinPacketQueue::QueuedPacket* packet);
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void OnPaddingSent(DataSize padding_sent);
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Timestamp CurrentTime() const;
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Clock* const clock_;
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PacketSender* const packet_sender_;
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const std::unique_ptr<FieldTrialBasedConfig> fallback_field_trials_;
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const WebRtcKeyValueConfig* field_trials_;
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const bool drain_large_queues_;
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const bool send_padding_if_silent_;
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const bool pace_audio_;
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TimeDelta min_packet_limit_;
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// TODO(webrtc:9716): Remove this when we are certain clocks are monotonic.
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// The last millisecond timestamp returned by |clock_|.
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mutable Timestamp last_timestamp_;
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bool paused_;
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// This is the media budget, keeping track of how many bits of media
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// we can pace out during the current interval.
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IntervalBudget media_budget_;
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// This is the padding budget, keeping track of how many bits of padding we're
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// allowed to send out during the current interval. This budget will be
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// utilized when there's no media to send.
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IntervalBudget padding_budget_;
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BitrateProber prober_;
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bool probing_send_failure_;
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bool padding_failure_state_;
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DataRate pacing_bitrate_;
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Timestamp time_last_process_;
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Timestamp last_send_time_;
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absl::optional<Timestamp> first_sent_packet_time_;
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RoundRobinPacketQueue packet_queue_;
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uint64_t packet_counter_;
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DataSize congestion_window_size_;
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DataSize outstanding_data_;
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TimeDelta queue_time_limit;
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bool account_for_audio_;
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// If true, PacedSender should only reference packets as in legacy mode.
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// If false, PacedSender may have direct ownership of RtpPacketToSend objects.
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// Defaults to true, will be changed to default false soon.
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const bool legacy_packet_referencing_;
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};
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} // namespace webrtc
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#endif // MODULES_PACING_PACING_CONTROLLER_H_
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